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android_kernel_xiaomi_sm8350/sound/soc/codecs/uda1380.c
Mark Brown dee89c4d94 ASoC: Merge snd_soc_ops into snd_soc_dai_ops 
Liam Girdwood's ASoC v2 work avoids having two different ops structures
for DAIs by merging the members of struct snd_soc_ops into struct
snd_soc_dai_ops, allowing per DAI configuration for everything.
Backport this change.

This paves the way for future work allowing any combination of DAIs to
be connected rather than having fixed purpose CODEC and CPU DAIs and
only allowing CODEC<->CPU interconnections.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-11-21 14:12:10 +00:00

847 lines
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/*
* uda1380.c - Philips UDA1380 ALSA SoC audio driver
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License version 2 as
* published by the Free Software Foundation.
*
* Copyright (c) 2007 Philipp Zabel <philipp.zabel@gmail.com>
* Improved support for DAPM and audio routing/mixing capabilities,
* added TLV support.
*
* Modified by Richard Purdie <richard@openedhand.com> to fit into SoC
* codec model.
*
* Copyright (c) 2005 Giorgio Padrin <giorgio@mandarinlogiq.org>
* Copyright 2005 Openedhand Ltd.
*/
#include <linux/module.h>
#include <linux/init.h>
#include <linux/types.h>
#include <linux/string.h>
#include <linux/slab.h>
#include <linux/errno.h>
#include <linux/ioctl.h>
#include <linux/delay.h>
#include <linux/i2c.h>
#include <sound/core.h>
#include <sound/control.h>
#include <sound/initval.h>
#include <sound/info.h>
#include <sound/soc.h>
#include <sound/soc-dapm.h>
#include <sound/tlv.h>
#include "uda1380.h"
#define UDA1380_VERSION "0.6"
/*
* uda1380 register cache
*/
static const u16 uda1380_reg[UDA1380_CACHEREGNUM] = {
0x0502, 0x0000, 0x0000, 0x3f3f,
0x0202, 0x0000, 0x0000, 0x0000,
0x0000, 0x0000, 0x0000, 0x0000,
0x0000, 0x0000, 0x0000, 0x0000,
0x0000, 0xff00, 0x0000, 0x4800,
0x0000, 0x0000, 0x0000, 0x0000,
0x0000, 0x0000, 0x0000, 0x0000,
0x0000, 0x0000, 0x0000, 0x0000,
0x0000, 0x8000, 0x0002, 0x0000,
};
/*
* read uda1380 register cache
*/
static inline unsigned int uda1380_read_reg_cache(struct snd_soc_codec *codec,
unsigned int reg)
{
u16 *cache = codec->reg_cache;
if (reg == UDA1380_RESET)
return 0;
if (reg >= UDA1380_CACHEREGNUM)
return -1;
return cache[reg];
}
/*
* write uda1380 register cache
*/
static inline void uda1380_write_reg_cache(struct snd_soc_codec *codec,
u16 reg, unsigned int value)
{
u16 *cache = codec->reg_cache;
if (reg >= UDA1380_CACHEREGNUM)
return;
cache[reg] = value;
}
/*
* write to the UDA1380 register space
*/
static int uda1380_write(struct snd_soc_codec *codec, unsigned int reg,
unsigned int value)
{
u8 data[3];
/* data is
* data[0] is register offset
* data[1] is MS byte
* data[2] is LS byte
*/
data[0] = reg;
data[1] = (value & 0xff00) >> 8;
data[2] = value & 0x00ff;
uda1380_write_reg_cache(codec, reg, value);
/* the interpolator & decimator regs must only be written when the
* codec DAI is active.
*/
if (!codec->active && (reg >= UDA1380_MVOL))
return 0;
pr_debug("uda1380: hw write %x val %x\n", reg, value);
if (codec->hw_write(codec->control_data, data, 3) == 3) {
unsigned int val;
i2c_master_send(codec->control_data, data, 1);
i2c_master_recv(codec->control_data, data, 2);
val = (data[0]<<8) | data[1];
if (val != value) {
pr_debug("uda1380: READ BACK VAL %x\n",
(data[0]<<8) | data[1]);
return -EIO;
}
return 0;
} else
return -EIO;
}
#define uda1380_reset(c) uda1380_write(c, UDA1380_RESET, 0)
/* declarations of ALSA reg_elem_REAL controls */
static const char *uda1380_deemp[] = {
"None",
"32kHz",
"44.1kHz",
"48kHz",
"96kHz",
};
static const char *uda1380_input_sel[] = {
"Line",
"Mic + Line R",
"Line L",
"Mic",
};
static const char *uda1380_output_sel[] = {
"DAC",
"Analog Mixer",
};
static const char *uda1380_spf_mode[] = {
"Flat",
"Minimum1",
"Minimum2",
"Maximum"
};
static const char *uda1380_capture_sel[] = {
"ADC",
"Digital Mixer"
};
static const char *uda1380_sel_ns[] = {
"3rd-order",
"5th-order"
};
static const char *uda1380_mix_control[] = {
"off",
"PCM only",
"before sound processing",
"after sound processing"
};
static const char *uda1380_sdet_setting[] = {
"3200",
"4800",
"9600",
"19200"
};
static const char *uda1380_os_setting[] = {
"single-speed",
"double-speed (no mixing)",
"quad-speed (no mixing)"
};
static const struct soc_enum uda1380_deemp_enum[] = {
SOC_ENUM_SINGLE(UDA1380_DEEMP, 8, 5, uda1380_deemp),
SOC_ENUM_SINGLE(UDA1380_DEEMP, 0, 5, uda1380_deemp),
};
static const struct soc_enum uda1380_input_sel_enum =
SOC_ENUM_SINGLE(UDA1380_ADC, 2, 4, uda1380_input_sel); /* SEL_MIC, SEL_LNA */
static const struct soc_enum uda1380_output_sel_enum =
SOC_ENUM_SINGLE(UDA1380_PM, 7, 2, uda1380_output_sel); /* R02_EN_AVC */
static const struct soc_enum uda1380_spf_enum =
SOC_ENUM_SINGLE(UDA1380_MODE, 14, 4, uda1380_spf_mode); /* M */
static const struct soc_enum uda1380_capture_sel_enum =
SOC_ENUM_SINGLE(UDA1380_IFACE, 6, 2, uda1380_capture_sel); /* SEL_SOURCE */
static const struct soc_enum uda1380_sel_ns_enum =
SOC_ENUM_SINGLE(UDA1380_MIXER, 14, 2, uda1380_sel_ns); /* SEL_NS */
static const struct soc_enum uda1380_mix_enum =
SOC_ENUM_SINGLE(UDA1380_MIXER, 12, 4, uda1380_mix_control); /* MIX, MIX_POS */
static const struct soc_enum uda1380_sdet_enum =
SOC_ENUM_SINGLE(UDA1380_MIXER, 4, 4, uda1380_sdet_setting); /* SD_VALUE */
static const struct soc_enum uda1380_os_enum =
SOC_ENUM_SINGLE(UDA1380_MIXER, 0, 3, uda1380_os_setting); /* OS */
/*
* from -48 dB in 1.5 dB steps (mute instead of -49.5 dB)
*/
static DECLARE_TLV_DB_SCALE(amix_tlv, -4950, 150, 1);
/*
* from -78 dB in 1 dB steps (3 dB steps, really. LSB are ignored),
* from -66 dB in 0.5 dB steps (2 dB steps, really) and
* from -52 dB in 0.25 dB steps
*/
static const unsigned int mvol_tlv[] = {
TLV_DB_RANGE_HEAD(3),
0, 15, TLV_DB_SCALE_ITEM(-8200, 100, 1),
16, 43, TLV_DB_SCALE_ITEM(-6600, 50, 0),
44, 252, TLV_DB_SCALE_ITEM(-5200, 25, 0),
};
/*
* from -72 dB in 1.5 dB steps (6 dB steps really),
* from -66 dB in 0.75 dB steps (3 dB steps really),
* from -60 dB in 0.5 dB steps (2 dB steps really) and
* from -46 dB in 0.25 dB steps
*/
static const unsigned int vc_tlv[] = {
TLV_DB_RANGE_HEAD(4),
0, 7, TLV_DB_SCALE_ITEM(-7800, 150, 1),
8, 15, TLV_DB_SCALE_ITEM(-6600, 75, 0),
16, 43, TLV_DB_SCALE_ITEM(-6000, 50, 0),
44, 228, TLV_DB_SCALE_ITEM(-4600, 25, 0),
};
/* from 0 to 6 dB in 2 dB steps if SPF mode != flat */
static DECLARE_TLV_DB_SCALE(tr_tlv, 0, 200, 0);
/* from 0 to 24 dB in 2 dB steps, if SPF mode == maximum, otherwise cuts
* off at 18 dB max) */
static DECLARE_TLV_DB_SCALE(bb_tlv, 0, 200, 0);
/* from -63 to 24 dB in 0.5 dB steps (-128...48) */
static DECLARE_TLV_DB_SCALE(dec_tlv, -6400, 50, 1);
/* from 0 to 24 dB in 3 dB steps */
static DECLARE_TLV_DB_SCALE(pga_tlv, 0, 300, 0);
/* from 0 to 30 dB in 2 dB steps */
static DECLARE_TLV_DB_SCALE(vga_tlv, 0, 200, 0);
static const struct snd_kcontrol_new uda1380_snd_controls[] = {
SOC_DOUBLE_TLV("Analog Mixer Volume", UDA1380_AMIX, 0, 8, 44, 1, amix_tlv), /* AVCR, AVCL */
SOC_DOUBLE_TLV("Master Playback Volume", UDA1380_MVOL, 0, 8, 252, 1, mvol_tlv), /* MVCL, MVCR */
SOC_SINGLE_TLV("ADC Playback Volume", UDA1380_MIXVOL, 8, 228, 1, vc_tlv), /* VC2 */
SOC_SINGLE_TLV("PCM Playback Volume", UDA1380_MIXVOL, 0, 228, 1, vc_tlv), /* VC1 */
SOC_ENUM("Sound Processing Filter", uda1380_spf_enum), /* M */
SOC_DOUBLE_TLV("Tone Control - Treble", UDA1380_MODE, 4, 12, 3, 0, tr_tlv), /* TRL, TRR */
SOC_DOUBLE_TLV("Tone Control - Bass", UDA1380_MODE, 0, 8, 15, 0, bb_tlv), /* BBL, BBR */
/**/ SOC_SINGLE("Master Playback Switch", UDA1380_DEEMP, 14, 1, 1), /* MTM */
SOC_SINGLE("ADC Playback Switch", UDA1380_DEEMP, 11, 1, 1), /* MT2 from decimation filter */
SOC_ENUM("ADC Playback De-emphasis", uda1380_deemp_enum[0]), /* DE2 */
SOC_SINGLE("PCM Playback Switch", UDA1380_DEEMP, 3, 1, 1), /* MT1, from digital data input */
SOC_ENUM("PCM Playback De-emphasis", uda1380_deemp_enum[1]), /* DE1 */
SOC_SINGLE("DAC Polarity inverting Switch", UDA1380_MIXER, 15, 1, 0), /* DA_POL_INV */
SOC_ENUM("Noise Shaper", uda1380_sel_ns_enum), /* SEL_NS */
SOC_ENUM("Digital Mixer Signal Control", uda1380_mix_enum), /* MIX_POS, MIX */
SOC_SINGLE("Silence Switch", UDA1380_MIXER, 7, 1, 0), /* SILENCE, force DAC output to silence */
SOC_SINGLE("Silence Detector Switch", UDA1380_MIXER, 6, 1, 0), /* SDET_ON */
SOC_ENUM("Silence Detector Setting", uda1380_sdet_enum), /* SD_VALUE */
SOC_ENUM("Oversampling Input", uda1380_os_enum), /* OS */
SOC_DOUBLE_S8_TLV("ADC Capture Volume", UDA1380_DEC, -128, 48, dec_tlv), /* ML_DEC, MR_DEC */
/**/ SOC_SINGLE("ADC Capture Switch", UDA1380_PGA, 15, 1, 1), /* MT_ADC */
SOC_DOUBLE_TLV("Line Capture Volume", UDA1380_PGA, 0, 8, 8, 0, pga_tlv), /* PGA_GAINCTRLL, PGA_GAINCTRLR */
SOC_SINGLE("ADC Polarity inverting Switch", UDA1380_ADC, 12, 1, 0), /* ADCPOL_INV */
SOC_SINGLE_TLV("Mic Capture Volume", UDA1380_ADC, 8, 15, 0, vga_tlv), /* VGA_CTRL */
SOC_SINGLE("DC Filter Bypass Switch", UDA1380_ADC, 1, 1, 0), /* SKIP_DCFIL (before decimator) */
SOC_SINGLE("DC Filter Enable Switch", UDA1380_ADC, 0, 1, 0), /* EN_DCFIL (at output of decimator) */
SOC_SINGLE("AGC Timing", UDA1380_AGC, 8, 7, 0), /* TODO: enum, see table 62 */
SOC_SINGLE("AGC Target level", UDA1380_AGC, 2, 3, 1), /* AGC_LEVEL */
/* -5.5, -8, -11.5, -14 dBFS */
SOC_SINGLE("AGC Switch", UDA1380_AGC, 0, 1, 0),
};
/* add non dapm controls */
static int uda1380_add_controls(struct snd_soc_codec *codec)
{
int err, i;
for (i = 0; i < ARRAY_SIZE(uda1380_snd_controls); i++) {
err = snd_ctl_add(codec->card,
snd_soc_cnew(&uda1380_snd_controls[i], codec, NULL));
if (err < 0)
return err;
}
return 0;
}
/* Input mux */
static const struct snd_kcontrol_new uda1380_input_mux_control =
SOC_DAPM_ENUM("Route", uda1380_input_sel_enum);
/* Output mux */
static const struct snd_kcontrol_new uda1380_output_mux_control =
SOC_DAPM_ENUM("Route", uda1380_output_sel_enum);
/* Capture mux */
static const struct snd_kcontrol_new uda1380_capture_mux_control =
SOC_DAPM_ENUM("Route", uda1380_capture_sel_enum);
static const struct snd_soc_dapm_widget uda1380_dapm_widgets[] = {
SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, 0, 0,
&uda1380_input_mux_control),
SND_SOC_DAPM_MUX("Output Mux", SND_SOC_NOPM, 0, 0,
&uda1380_output_mux_control),
SND_SOC_DAPM_MUX("Capture Mux", SND_SOC_NOPM, 0, 0,
&uda1380_capture_mux_control),
SND_SOC_DAPM_PGA("Left PGA", UDA1380_PM, 3, 0, NULL, 0),
SND_SOC_DAPM_PGA("Right PGA", UDA1380_PM, 1, 0, NULL, 0),
SND_SOC_DAPM_PGA("Mic LNA", UDA1380_PM, 4, 0, NULL, 0),
SND_SOC_DAPM_ADC("Left ADC", "Left Capture", UDA1380_PM, 2, 0),
SND_SOC_DAPM_ADC("Right ADC", "Right Capture", UDA1380_PM, 0, 0),
SND_SOC_DAPM_INPUT("VINM"),
SND_SOC_DAPM_INPUT("VINL"),
SND_SOC_DAPM_INPUT("VINR"),
SND_SOC_DAPM_MIXER("Analog Mixer", UDA1380_PM, 6, 0, NULL, 0),
SND_SOC_DAPM_OUTPUT("VOUTLHP"),
SND_SOC_DAPM_OUTPUT("VOUTRHP"),
SND_SOC_DAPM_OUTPUT("VOUTL"),
SND_SOC_DAPM_OUTPUT("VOUTR"),
SND_SOC_DAPM_DAC("DAC", "Playback", UDA1380_PM, 10, 0),
SND_SOC_DAPM_PGA("HeadPhone Driver", UDA1380_PM, 13, 0, NULL, 0),
};
static const struct snd_soc_dapm_route audio_map[] = {
/* output mux */
{"HeadPhone Driver", NULL, "Output Mux"},
{"VOUTR", NULL, "Output Mux"},
{"VOUTL", NULL, "Output Mux"},
{"Analog Mixer", NULL, "VINR"},
{"Analog Mixer", NULL, "VINL"},
{"Analog Mixer", NULL, "DAC"},
{"Output Mux", "DAC", "DAC"},
{"Output Mux", "Analog Mixer", "Analog Mixer"},
/* {"DAC", "Digital Mixer", "I2S" } */
/* headphone driver */
{"VOUTLHP", NULL, "HeadPhone Driver"},
{"VOUTRHP", NULL, "HeadPhone Driver"},
/* input mux */
{"Left ADC", NULL, "Input Mux"},
{"Input Mux", "Mic", "Mic LNA"},
{"Input Mux", "Mic + Line R", "Mic LNA"},
{"Input Mux", "Line L", "Left PGA"},
{"Input Mux", "Line", "Left PGA"},
/* right input */
{"Right ADC", "Mic + Line R", "Right PGA"},
{"Right ADC", "Line", "Right PGA"},
/* inputs */
{"Mic LNA", NULL, "VINM"},
{"Left PGA", NULL, "VINL"},
{"Right PGA", NULL, "VINR"},
};
static int uda1380_add_widgets(struct snd_soc_codec *codec)
{
snd_soc_dapm_new_controls(codec, uda1380_dapm_widgets,
ARRAY_SIZE(uda1380_dapm_widgets));
snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
snd_soc_dapm_new_widgets(codec);
return 0;
}
static int uda1380_set_dai_fmt(struct snd_soc_dai *codec_dai,
unsigned int fmt)
{
struct snd_soc_codec *codec = codec_dai->codec;
int iface;
/* set up DAI based upon fmt */
iface = uda1380_read_reg_cache(codec, UDA1380_IFACE);
iface &= ~(R01_SFORI_MASK | R01_SIM | R01_SFORO_MASK);
/* FIXME: how to select I2S for DATAO and MSB for DATAI correctly? */
switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_I2S:
iface |= R01_SFORI_I2S | R01_SFORO_I2S;
break;
case SND_SOC_DAIFMT_LSB:
iface |= R01_SFORI_LSB16 | R01_SFORO_I2S;
break;
case SND_SOC_DAIFMT_MSB:
iface |= R01_SFORI_MSB | R01_SFORO_I2S;
}
if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) == SND_SOC_DAIFMT_CBM_CFM)
iface |= R01_SIM;
uda1380_write(codec, UDA1380_IFACE, iface);
return 0;
}
/*
* Flush reg cache
* We can only write the interpolator and decimator registers
* when the DAI is being clocked by the CPU DAI. It's up to the
* machine and cpu DAI driver to do this before we are called.
*/
static int uda1380_pcm_prepare(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
struct snd_soc_codec *codec = socdev->codec;
int reg, reg_start, reg_end, clk;
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
reg_start = UDA1380_MVOL;
reg_end = UDA1380_MIXER;
} else {
reg_start = UDA1380_DEC;
reg_end = UDA1380_AGC;
}
/* FIXME disable DAC_CLK */
clk = uda1380_read_reg_cache(codec, UDA1380_CLK);
uda1380_write(codec, UDA1380_CLK, clk & ~R00_DAC_CLK);
for (reg = reg_start; reg <= reg_end; reg++) {
pr_debug("uda1380: flush reg %x val %x:", reg,
uda1380_read_reg_cache(codec, reg));
uda1380_write(codec, reg, uda1380_read_reg_cache(codec, reg));
}
/* FIXME enable DAC_CLK */
uda1380_write(codec, UDA1380_CLK, clk | R00_DAC_CLK);
return 0;
}
static int uda1380_pcm_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
struct snd_soc_codec *codec = socdev->codec;
u16 clk = uda1380_read_reg_cache(codec, UDA1380_CLK);
/* set WSPLL power and divider if running from this clock */
if (clk & R00_DAC_CLK) {
int rate = params_rate(params);
u16 pm = uda1380_read_reg_cache(codec, UDA1380_PM);
clk &= ~0x3; /* clear SEL_LOOP_DIV */
switch (rate) {
case 6250 ... 12500:
clk |= 0x0;
break;
case 12501 ... 25000:
clk |= 0x1;
break;
case 25001 ... 50000:
clk |= 0x2;
break;
case 50001 ... 100000:
clk |= 0x3;
break;
}
uda1380_write(codec, UDA1380_PM, R02_PON_PLL | pm);
}
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
clk |= R00_EN_DAC | R00_EN_INT;
else
clk |= R00_EN_ADC | R00_EN_DEC;
uda1380_write(codec, UDA1380_CLK, clk);
return 0;
}
static void uda1380_pcm_shutdown(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
struct snd_soc_codec *codec = socdev->codec;
u16 clk = uda1380_read_reg_cache(codec, UDA1380_CLK);
/* shut down WSPLL power if running from this clock */
if (clk & R00_DAC_CLK) {
u16 pm = uda1380_read_reg_cache(codec, UDA1380_PM);
uda1380_write(codec, UDA1380_PM, ~R02_PON_PLL & pm);
}
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
clk &= ~(R00_EN_DAC | R00_EN_INT);
else
clk &= ~(R00_EN_ADC | R00_EN_DEC);
uda1380_write(codec, UDA1380_CLK, clk);
}
static int uda1380_mute(struct snd_soc_dai *codec_dai, int mute)
{
struct snd_soc_codec *codec = codec_dai->codec;
u16 mute_reg = uda1380_read_reg_cache(codec, UDA1380_DEEMP) & ~R13_MTM;
/* FIXME: mute(codec,0) is called when the magician clock is already
* set to WSPLL, but for some unknown reason writing to interpolator
* registers works only when clocked by SYSCLK */
u16 clk = uda1380_read_reg_cache(codec, UDA1380_CLK);
uda1380_write(codec, UDA1380_CLK, ~R00_DAC_CLK & clk);
if (mute)
uda1380_write(codec, UDA1380_DEEMP, mute_reg | R13_MTM);
else
uda1380_write(codec, UDA1380_DEEMP, mute_reg);
uda1380_write(codec, UDA1380_CLK, clk);
return 0;
}
static int uda1380_set_bias_level(struct snd_soc_codec *codec,
enum snd_soc_bias_level level)
{
int pm = uda1380_read_reg_cache(codec, UDA1380_PM);
switch (level) {
case SND_SOC_BIAS_ON:
case SND_SOC_BIAS_PREPARE:
uda1380_write(codec, UDA1380_PM, R02_PON_BIAS | pm);
break;
case SND_SOC_BIAS_STANDBY:
uda1380_write(codec, UDA1380_PM, R02_PON_BIAS);
break;
case SND_SOC_BIAS_OFF:
uda1380_write(codec, UDA1380_PM, 0x0);
break;
}
codec->bias_level = level;
return 0;
}
#define UDA1380_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\
SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000)
struct snd_soc_dai uda1380_dai[] = {
{
.name = "UDA1380",
.playback = {
.stream_name = "Playback",
.channels_min = 1,
.channels_max = 2,
.rates = UDA1380_RATES,
.formats = SNDRV_PCM_FMTBIT_S16_LE,},
.capture = {
.stream_name = "Capture",
.channels_min = 1,
.channels_max = 2,
.rates = UDA1380_RATES,
.formats = SNDRV_PCM_FMTBIT_S16_LE,},
.ops = {
.hw_params = uda1380_pcm_hw_params,
.shutdown = uda1380_pcm_shutdown,
.prepare = uda1380_pcm_prepare,
.digital_mute = uda1380_mute,
.set_fmt = uda1380_set_dai_fmt,
},
},
{ /* playback only - dual interface */
.name = "UDA1380",
.playback = {
.stream_name = "Playback",
.channels_min = 1,
.channels_max = 2,
.rates = UDA1380_RATES,
.formats = SNDRV_PCM_FMTBIT_S16_LE,
},
.ops = {
.hw_params = uda1380_pcm_hw_params,
.shutdown = uda1380_pcm_shutdown,
.prepare = uda1380_pcm_prepare,
.digital_mute = uda1380_mute,
.set_fmt = uda1380_set_dai_fmt,
},
},
{ /* capture only - dual interface*/
.name = "UDA1380",
.capture = {
.stream_name = "Capture",
.channels_min = 1,
.channels_max = 2,
.rates = UDA1380_RATES,
.formats = SNDRV_PCM_FMTBIT_S16_LE,
},
.ops = {
.hw_params = uda1380_pcm_hw_params,
.shutdown = uda1380_pcm_shutdown,
.prepare = uda1380_pcm_prepare,
.set_fmt = uda1380_set_dai_fmt,
},
},
};
EXPORT_SYMBOL_GPL(uda1380_dai);
static int uda1380_suspend(struct platform_device *pdev, pm_message_t state)
{
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
struct snd_soc_codec *codec = socdev->codec;
uda1380_set_bias_level(codec, SND_SOC_BIAS_OFF);
return 0;
}
static int uda1380_resume(struct platform_device *pdev)
{
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
struct snd_soc_codec *codec = socdev->codec;
int i;
u8 data[2];
u16 *cache = codec->reg_cache;
/* Sync reg_cache with the hardware */
for (i = 0; i < ARRAY_SIZE(uda1380_reg); i++) {
data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001);
data[1] = cache[i] & 0x00ff;
codec->hw_write(codec->control_data, data, 2);
}
uda1380_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
uda1380_set_bias_level(codec, codec->suspend_bias_level);
return 0;
}
/*
* initialise the UDA1380 driver
* register mixer and dsp interfaces with the kernel
*/
static int uda1380_init(struct snd_soc_device *socdev, int dac_clk)
{
struct snd_soc_codec *codec = socdev->codec;
int ret = 0;
codec->name = "UDA1380";
codec->owner = THIS_MODULE;
codec->read = uda1380_read_reg_cache;
codec->write = uda1380_write;
codec->set_bias_level = uda1380_set_bias_level;
codec->dai = uda1380_dai;
codec->num_dai = ARRAY_SIZE(uda1380_dai);
codec->reg_cache = kmemdup(uda1380_reg, sizeof(uda1380_reg),
GFP_KERNEL);
if (codec->reg_cache == NULL)
return -ENOMEM;
codec->reg_cache_size = ARRAY_SIZE(uda1380_reg);
codec->reg_cache_step = 1;
uda1380_reset(codec);
/* register pcms */
ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
if (ret < 0) {
pr_err("uda1380: failed to create pcms\n");
goto pcm_err;
}
/* power on device */
uda1380_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
/* set clock input */
switch (dac_clk) {
case UDA1380_DAC_CLK_SYSCLK:
uda1380_write(codec, UDA1380_CLK, 0);
break;
case UDA1380_DAC_CLK_WSPLL:
uda1380_write(codec, UDA1380_CLK, R00_DAC_CLK);
break;
}
/* uda1380 init */
uda1380_add_controls(codec);
uda1380_add_widgets(codec);
ret = snd_soc_register_card(socdev);
if (ret < 0) {
pr_err("uda1380: failed to register card\n");
goto card_err;
}
return ret;
card_err:
snd_soc_free_pcms(socdev);
snd_soc_dapm_free(socdev);
pcm_err:
kfree(codec->reg_cache);
return ret;
}
static struct snd_soc_device *uda1380_socdev;
#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
static int uda1380_i2c_probe(struct i2c_client *i2c,
const struct i2c_device_id *id)
{
struct snd_soc_device *socdev = uda1380_socdev;
struct uda1380_setup_data *setup = socdev->codec_data;
struct snd_soc_codec *codec = socdev->codec;
int ret;
i2c_set_clientdata(i2c, codec);
codec->control_data = i2c;
ret = uda1380_init(socdev, setup->dac_clk);
if (ret < 0)
pr_err("uda1380: failed to initialise UDA1380\n");
return ret;
}
static int uda1380_i2c_remove(struct i2c_client *client)
{
struct snd_soc_codec *codec = i2c_get_clientdata(client);
kfree(codec->reg_cache);
return 0;
}
static const struct i2c_device_id uda1380_i2c_id[] = {
{ "uda1380", 0 },
{ }
};
MODULE_DEVICE_TABLE(i2c, uda1380_i2c_id);
static struct i2c_driver uda1380_i2c_driver = {
.driver = {
.name = "UDA1380 I2C Codec",
.owner = THIS_MODULE,
},
.probe = uda1380_i2c_probe,
.remove = uda1380_i2c_remove,
.id_table = uda1380_i2c_id,
};
static int uda1380_add_i2c_device(struct platform_device *pdev,
const struct uda1380_setup_data *setup)
{
struct i2c_board_info info;
struct i2c_adapter *adapter;
struct i2c_client *client;
int ret;
ret = i2c_add_driver(&uda1380_i2c_driver);
if (ret != 0) {
dev_err(&pdev->dev, "can't add i2c driver\n");
return ret;
}
memset(&info, 0, sizeof(struct i2c_board_info));
info.addr = setup->i2c_address;
strlcpy(info.type, "uda1380", I2C_NAME_SIZE);
adapter = i2c_get_adapter(setup->i2c_bus);
if (!adapter) {
dev_err(&pdev->dev, "can't get i2c adapter %d\n",
setup->i2c_bus);
goto err_driver;
}
client = i2c_new_device(adapter, &info);
i2c_put_adapter(adapter);
if (!client) {
dev_err(&pdev->dev, "can't add i2c device at 0x%x\n",
(unsigned int)info.addr);
goto err_driver;
}
return 0;
err_driver:
i2c_del_driver(&uda1380_i2c_driver);
return -ENODEV;
}
#endif
static int uda1380_probe(struct platform_device *pdev)
{
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
struct uda1380_setup_data *setup;
struct snd_soc_codec *codec;
int ret;
pr_info("UDA1380 Audio Codec %s", UDA1380_VERSION);
setup = socdev->codec_data;
codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
if (codec == NULL)
return -ENOMEM;
socdev->codec = codec;
mutex_init(&codec->mutex);
INIT_LIST_HEAD(&codec->dapm_widgets);
INIT_LIST_HEAD(&codec->dapm_paths);
uda1380_socdev = socdev;
ret = -ENODEV;
#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
if (setup->i2c_address) {
codec->hw_write = (hw_write_t)i2c_master_send;
ret = uda1380_add_i2c_device(pdev, setup);
}
#endif
if (ret != 0)
kfree(codec);
return ret;
}
/* power down chip */
static int uda1380_remove(struct platform_device *pdev)
{
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
struct snd_soc_codec *codec = socdev->codec;
if (codec->control_data)
uda1380_set_bias_level(codec, SND_SOC_BIAS_OFF);
snd_soc_free_pcms(socdev);
snd_soc_dapm_free(socdev);
#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
i2c_unregister_device(codec->control_data);
i2c_del_driver(&uda1380_i2c_driver);
#endif
kfree(codec);
return 0;
}
struct snd_soc_codec_device soc_codec_dev_uda1380 = {
.probe = uda1380_probe,
.remove = uda1380_remove,
.suspend = uda1380_suspend,
.resume = uda1380_resume,
};
EXPORT_SYMBOL_GPL(soc_codec_dev_uda1380);
MODULE_AUTHOR("Giorgio Padrin");
MODULE_DESCRIPTION("Audio support for codec Philips UDA1380");
MODULE_LICENSE("GPL");
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